The present invention relates generally to telephony and more particularly to call processing for Voice Over IP (VoIP) devices and processing of other media sessions using SIP devices. VoIP devices allow telephony services to be provided over packet switched networks such as the Internet and other data networks, and thus provide flexibility that cannot be achieved by plain old telephone system (POTS) legacy telephones that use circuit switched systems such as the public switched telephone network (PSTN). Rather than connecting a VoIP device to a fixed telephone line, the device can be connected to the Internet at any location. In addition, VoIP devices offer potential cost savings for voice calls, particularly for long distance calls, lower overall network costs for businesses since voice and data are carried over the same network, and flexibility in relocating phones. VoIP devices include so-called SIP phones that communicate according to the IETF standard session initiation protocol (SIP) and other devices such as those employing the ITU standard H.323 protocol. Because these devices communicate via packet-switched networks, calls placed to a VoIP device must specify a universal resource identifier (URI) such as a SIP URI to allow proper routing of the call packets through the network. In addition to voice calls, SIP devices and other VoIP devices may support other types of sessions, such as multimedia distribution, multimedia conferences, etc., where the SIP protocol is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants.
One shortcoming of SIP phones and VoIP devices generally is that persons wishing to call a SIP device must know the URI (e.g., SIP URI) of the device they are calling or a legacy phone number assigned to the device. The URI is generally in the form user1@VoIP-provider-1.com like an Email address, wherein SIP, Email and other IP applications use URI for addressing purposes. The SIP URI and any legacy phone number are assigned by the subscriber's VoIP service provider, and accordingly change whenever the user switches to a new provider. When a VoIP user wants to receive calls from friends, family, colleagues, etc., he or she must advertise his or her SIP URI and/or the legacy number to each such person, which can take a lot of time. This problem is similar to the situation where a person gets a brand new telephone number for a legacy or mobile phone. For legacy telephones, so-called “Number Portability” techniques and systems have been developed to allow persons to retain their old legacy telephone number even though they change service providers. However, there is currently no analogous solution providing VoIP URI portability and number portability is not common outside the United States. Another problem with conventional VoIP services is that the VoIP device needs to be assigned a legacy phone number, in addition to the SIP URI, so that voice calls can be received from legacy callers via the PSTN. PSTN phone numbers are scarce and expensive resources, and the “Number Portability” and other regulatory features that go with them are expensive and cumbersome. Thus, there remains a need for systems and methods by which VoIP device users could use their existing and well-known Email address as a SIP URI, and do not need to update all preferred calling parties with new URIs and/or legacy phone numbers whenever they switch VoIP phone service providers.